This invention relates in general to the processing and transmission of electrical signals and in particular to circuits for converting analog signals to digital signals and then recovering the analog signals from the digital signals.
Many digital systems have been proposed for transmitting electrical signals such as in the delivery of audio signals from a broadcast center to the consumer. One proposed system employs linear pulse code modulation (PCM) in which the quantizing step-sizes remain constant throughout the transmission. To preserve the quality of signals transmitted and in particular to retain the resolution for small signals, at least 13 binary bits per sample are required, while 14 to 16 bit systems are commonly used. For broadcasting of audio signals, the bandwidth is typically 15 kHz. Thus linear PCM systems require high bit rates of transmission (a minimum of about 32 kHz.times.13 bits i.e. 416 kbit per second even without any provision for error identification or correction). A linear PCM system must also employ high precision components and is therefore expensive.
To reduce transmission bit rate, some conventional PCM systems have employed digital companding. While the transmission bit rate can be reduced somewhat, precision 13 or 14 bit converters must still be used and digital companding further adds complexity and cost to the circuitry. Hence linear or digitally companded PCM systems are unattractive for cost sensitive applications such as use in the home.
Delta modulation systems are attractive for signal transmission because of cost savings in their hardware; delta modulation systems employ simple circuits and do not require close component tolerance. Delta modulation systems are also inherently less disturbed by uncorrected transmission errors. Linear delta modulation systems however require much higher bit rates than PCM systems to achieve the same quality transmission and are therefore unattractive.
Bit-rate reduction has also been applied to delta modulation systems by varying the effective quantizing step-size. Such delta modulation systems are known as adaptive delta modulation (ADM) systems. Digital companding in multi-level PCM systems is performed usually by quantizing to the highest degree of resolution for small signals and then reducing the resolution for large amplitude signals. With ADM systems whose bit-streams have only two states, the adaptation is usually performed by directly altering the quantizing increment between those states either on an instantaneous (sample to sample) basis, on a block basis ("near-instantaneous") or at a syllabic rate.
In ADM, the adaptation can be considered equivalent to multiplication or division of the audio signal by the step-size. It is well known from modulation theory that the act of multiplying or dividing an analog signal by another signal will cause the resulting product signal to include modulation sidebands which contaminate the original analog signal. In an ADM encoder the digital output is a representation of this product signal. In order to eliminate these modulation sidebands and to reconstruct the original audio signal from the digital representation, a complementary process must be performed in the adaptive delta-demodulation, that is another multiplication or division. It is apparent that to reconstruct the audio perfectly, the multipliers or dividers in the encoder and decoder need to be `perfect` (or at least identically imperfect) and the multiplicative signals must also be precisely equal. The mismatch inherent in real circuits will result in the reconstructed audio having a spectrum differing from the original. The frequencies of the spectral difference will depend on the spectrum of the modulating or control signals, and the amplitude of the differences will depend on the magnitude of the circuit mismatches.
In ADM systems employing instantaneous or near-instantaneous adaptation, the adaptation is usually output controlled (operating from the output bit stream). The control signals of such systems have spectral components ranging from D.C. to above the sampling frequency so that the systems may have acceptable performance for coding transient signals. However, during digital transmission, some individual control bits of such control signals will become much more significant than other bits, so that a small percentage of errors in transmitting the wide band control signals (caused for example by a random or burst error) which happen to hit these critical control bits will cause the received control signals to deviate significantly from their proper values. Because of such errors, the magnitude of the spurious spectral components in the received audio can become very large even if the circuits are perfect.
It is a characteristic of human hearing that spurious spectral information is much less audible if it is close in frequency to the desired audio signal. If the spurious energy lies far from the desired audio signal it is much more likely to be audible.
With an ADM system, the tolerance of the control signal tracking (encoder and decoder) and the multiplier precision can be relaxed somewhat if the spectrum of the control signal is constrained to contain only low frequency information. This is because a low frequency modulating signal produces new frequencies which are close to the original audio frequencies, and we can tolerate some of these new frequencies in the decoded output since they will be masked by the audio signal.
Adaptation with relatively slow control signals is known as syllabic adaptation. A syllabic ADM system is much more appropriate for low cost high quality audio applications for the reasons mentioned above. Such a system is also more resistant to errors in transmission of the control signal since the audible effects will be less. However, reduction of control information bandwidth will cause the step-size control signal to be delayed relative to the analog input signal. Thus, the analog input signal will reach the A-D converter before the converter has completed or even started its adaptation in accordance with the control signal. In a high quality audio system, the bandwidth of the control signal cannot be narrower than a few kHz without causing noticeable or even intolerable distortion in the reproduced audio.
Syllabically adaptive delta modulation modifies the quantizing step-size to accommodate the time derivative or slope of the incoming signal. In an ideal syllabic ADM design, the step-size at each instant will be only slightly greater than the minimum needed thereby giving a minimum quantizing error and the best signal to noise ratio. However, to achieve comparable quality, even an ideal ADM system requires higher bit rates than a PCM system. If the bit rate of an ADM system is reduced to one comparable with that required for a PCM system, the noise modulation is unacceptable. The audibility of noise modulation accompanying low and middle frequency audio signals can be reduced by the use of fixed pre-emphasis and de-emphasis. However, this is done only at the expense of increased noise modulation when the audio signal contains significant high frequencies. The increased noise modulation is particularly disturbing for low frequency noise in the presence of predominantly high frequency signals.
Conventional syllabically adaptive delta mdoulation systems with limited adapting bandwidth (slow response time) suffer from transient distortion. Because of the limited adapting bandwidth the system is in slope overload during the time required to respond to a sudden increase in signal level. While ADM systems which suffer brief transient distortions may be acceptable for communication purposes since speech intelligibility may be unaffected, such distortions cannot be tolerated for many applications, such as high quality audio.